In typical telecommunications systems, voice calls and data are transmitted by carriers from one network to another network. Networks for transmitting voice calls include packet-switched networks transmitting calls using voice over Internet Protocols (VoIP), circuit-switched networks like the public switched telephone network (PSTN), asynchronous transfer mode (ATM) networks, etc. Recently, voice over packet (VOP) networks are becoming more widely deployed. Many incumbent local exchange and long-distance service providers use VoIP technology in the backhaul of their networks without the end user being aware that VoIP is involved.
Traditional service providers use techniques to manage service quality developed over the last 100 or more years for circuit-switched networks. Methods include tracking of customer and network trouble reports and re-design of voice networks. Service providers use well-understood rules to characterize service level in terms of voice quality (e.g., based on loss, delay, and echo), and in difficultly in establishing a call. Then, a service provider's main tool to assess service quality while the network is in operation is based on trouble reports from users, as well as general network equipment failure notification.
Voice quality is traditionally thought of as the end user's perception of quality. Network performance will affect voice quality. However, as VoIP technology increases in demand on a network and networks become more complicated with connections through the Internet and PSTN using IP phones (wired and wireless) and residential voice gateways, VoIP providers have a much more difficult time assuring the voice quality for their subscribers. Reasons for this include lack of control over the underlying transport network, such as when a service provider providing voice service from a residential gateway attaches to another provider's residential broadband cable modem or DSL (Digital Subscriber Line) service and the use of transport technology that can vary in quality. For example, using WLAN (wireless local area network) media to transport VoIP, especially when the wireless end user is moving between WLANs.
An example of networks and components for a VoIP call is illustrated in FIG. 1. Access network 10 could be any network accessing the Internet such as an IP, Asychronous Transfer Mode (ATM), or Ethernet network, which is a managed broadband network. Network 10 comprises a router 14 connected to various customer premise equipment and to media gateway 12. Media gateway 12 must be capable of detecting changing resource or network conditions. The ability to detect and monitor changing resource and network conditions can result in significant cost reductions and/or improved quality. Router 14 is connected to Internet Access Device (IAD) 16, wireless access point (AP) 22, and/or IP PBX (personal branch exchange) 32. A voice call may be placed between any of the customer equipment phones 18 connected to IAD 16, wireless IP phone 24 connected to AP 22, or IP PBX phone 30. Using special software, calls could also be placed through computer 20 connected to IAD 16 or portable computer 26 connected to AP 22.
Customer equipment is connected through access broadband network 10 to the Internet 34 by media gateway 12. On the far end is the PSTN 48, networking to POTS phone 52 through a Central Office 50. PSTN is also connected to the Internet 34 through a trunk gateway, composed of signal gateway 44, media gateway controller/proxy (MGC) 42, and trunk media gateway (MG) 46. IP and packet data (e.g., real time protocol (RTP packet data)) associated with the call is routed between IAD 16 and trunk MG 46. The trunk gateway system provides real-time two-way communications interfaces between the IP network (e.g., the Internet) and the PSTN 50. As another example, a VoIP call could be initiated between WIPP 24 and WIPP 40 connected to AP 38. In this call, voice signals and associated packet data are sent between MG 12 and MG 52 through Internet 42, thereby bypassing the PSTN 48 altogether.
Factors that affect voice quality in a VoIP network are fairly well understood. The level of control over these factors will vary from network to network. This is highlighted by the differences between a well-managed small network enterprise verses an unmanaged network such as the Internet. Network operational issues affect network performance and will create conditions that affect voice quality. These issues include outages/failures of network switches, routers, and bridges; outages/failure of VoIP elements such as call servers and gateways; and traffic management during peak periods and virus/denial of service attacks.
Software for VoIP systems is a critical ingredient of high-quality VoIP systems. There are many features that must be implemented for carrier-class systems. The most important software features include echo cancellation, voice compression, packet play-out software, tone processing, fax and modem support, packetization, signaling support, and network management. New networking technologies and deployment models are also causing additional challenges that affect the ability of VoIP service providers to guarantee the highest levels of service quality (e.g., toll quality) in their deployments. Two such examples are where the VoIP service provider does not control the underlying packet transport network, and the use of packet networks with potentially high delay and loss, such as in 802.11 WLAN (Wireless Local Area Network) technology.
The ability to detect and report on events in a network that adversely affect voice quality is critical for managing a voice network. The oldest network voice quality tool is the listening opinion tests, where human listeners rate call quality in a controlled setting (from ITU-T Spec. P.800). Overall results are compiled to produce a mean opinion score (MOS), which is based on a panel of listeners ranking the quality of a series of call samples on a scale of 1 (Bad) to 5 (Excellent). An aggregate score of 4 or more is considered toll quality, which is the standard for the PSTN. While this test has the disadvantage of being subjective, expensive, and time-consuming to produce, it is traditionally recognized as the most consistent measure of voice quality available.
Most of the subsequent voice quality measurement tools have involved algorithms and tools that can objectively measure voice quality. These are based on mathematical calculations on sound samples, rather than listening tests. In general, such tests can be roughly classified as active (or intrusive) and passive (or non-intrusive). Active tests perform calculations on test or simulated calls and thus intrude on normal network usage, while passive tests can perform calculations on active calls in live networks without any interruption of service
It is costly to test the quality of voice networks at the component and system level and to measure the performance of active networks, since revenue-producing traffic must be interrupted to perform the tests. Further, while testing algorithms can quantify deficiencies in speech quality, they do not produce information to help localize and identify the root causes of the situations causing the deficiency. Passive tests run in live networks without interrupting active calls and often use statistics gathered on active calls. The testing modules are actually embedded into the VoIP equipment at the use site and in the VoIP service provider's network.
In current VOP deployments, voice quality issues are first typically discovered and reported by customers which triggers an investigation and debugging by service providers. This method of problem detection can lead to longer problem resolution times and increase customer dissatisfaction. Currently, there exits no system and method that provides an enhanced means for service providers to effectively monitor their networks for potential voice quality issues and proactively isolate problems before customer complaints are received.